This is one of the most common questions we receive from our prospective customers. VoIP calls really do not require much bandwidth at all, and there are adjustments that can be made to condense the requirement even further.
The first mental hurdle to overcome is that your concurrent voice calls are no longer restricted to the number of analog “lines” you have provisioned for your company. With VoIP, your only restriction is the amount of bandwidth you have. Even if you are flooded with concurrent calls, many VoIP systems will capture the call and deliver it to your company voicemail, thus your customers will never hear a busy signal. However, the above scenario makes the question of bandwidth requirements very appropriate.
A single VoIP call requires 80k of bandwidth in a non-compressed environment. Thus, 10 concurrent calls active on your system would require 800K of bandwidth. However, it is common practice to utilize voice compression to reduce this requirement. A compression codec will lower the requirement to as little as 25k of bandwidth per call. Utilizing compression can help conserve bandwidth without a material sacrifice of voice quality, especially if your VoIP provider provides a Quality of Service software with their service.
There are some things you can do to test your company’s readiness for VoIP phone service, however a good VoIP service provider should consult with you prior to your installation on these very topics:
Run a speed test – run a free speed test of your bandwidth. Record the download and upload speeds, and try again later in the day and repeat every few days. You’re looking for your download speeds to be robust enough to handle your voice and data volume and for the speed results consistent as well.
Run an extended ping test – you should ping the potential service provider to check the latency between you and their server. You’re looking for the result to be under 110 milliseconds at a minimum. Ideally, a good circuit should produce latency in the 20 milliseconds or less range. Furthermore, it should be relatively stable. Results bouncing around (like 20, 20, 150, 90, 20, 20, 150, 20, etc) will produce packet drops and jittery phone calls.
Check for packet loss – a stable internet connection should experience no packet loss. VoIP service is dependent upon a no to very low packet loss. If packet loss is approaching 2%, contact your internet provider to service the line.
Determine your concurrent call volume – this may be via observation or through an understanding at which point you receive busy signals on your analog system. If you have five incoming lines, and you or your customers have experienced busy signals, then your concurrent call maximum is greater than five.
Do the math – Maximum concurrent call volume multiplied by 80k must be less than the lower of the upload or download speeds recorded on your speed test. We usually recommend that the maximum voice call packet transfer doesn’t absorb more than 40-50% of your bandwidth availability. However, this is entirely dependent upon your data requirements. Additionally, compressed codec could be deployed before considering more bandwidth.
The amount of bandwidth you have isn't the only variable in delivering quality phone conversations with VoIP. Many providers "solve" voice quality issues by oversizing bandwidth. However, there are other, more reliable solutions. Give sipVine a call today for a consultation. We're happy to help.
